similar to: Converting MP3 files to wav for Asterisk

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

I am trying to convert some a few WAV files to ogg format (using vorbis-tools-1.0beta4 oggenc). I tried: $ oggenc -o nash.ogg nash.wav ERROR: Wav file is unsupported subformat (must be 16 bit PCM or floating point PCM ERROR: Input file "nash.wav" is not a supported format And: splay -d - nash.wav | oggenc -r -o nash.ogg - The resulting file when played with ogg123 played so

Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Its the way some files are encoded. Is there a way I can make it skip the files that can be played? I use the

hi, i want convert mixmonitor recorded speech audio from wav to mp3 or aac can you recommend your settings for speech audio? filters, noise elimination, compression ratio, ... i will probably use lame thank you -- --------------------------------------- Marek Cervenka =======================================

Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or

Ok, Here's another bizarre one (no strange curve balls to throw this time :P). I have several mp3 files of some easy listening music that I pulled off some CDs we have. They sounds fine and are at a nice volume level. When I run this script I wrote: (I run it by doing ./script < filename) mpg123 -s --rate 44100 --mono $1.mp3 > $1.raw sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav

Hi, Would anyone have a clue on this issue? I'm running asterisk 1.4.13. Trying to get a WAV voicemail file attachment sent to my email address. Voicemail is working fine. Email notification of a new message works fine. However, when I set up voicemail.conf to have an attachment of the file sent to email I get an error message like this on the console: [Feb 17 13:33:37]

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

I cannot see my post with a patch for FLAC.sln (too long?) Here I repeat this message, with patch file compressed to .zip: --------------------------------------------------------------- Every time I open current FLAC.sln in MSVS 2005, it rewrites (fixes) this file. The diff between new and old files is attached. Current version has several problems: a) it has one extra EndProject line. b) it

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. - Andrew

By the way, I was delighted to find out that I could join .ogg files with cat, i.e. cat {file1.ogg} file2.ogg > outputfile.ogg <p>I successfully used this capability, together with oggenc and sox, to join two .WAV files in the following manner: <p>oggenc -q9 file1.wav ... oggenc -q9 file2.wav --- cat file1.ogg file2.ogg > outfile.ogg ox outfile.ogg outfile.wav I get